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Introduction

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• Advanced
  • Audio codecs: enables the audio codecs supported by the trunk and the ptime values: 20ms – 40ms – 60ms (20ms by default); in case a different priority is needed, use  “Set” -> “Codec” Dialplan application
  • Video codecs: enables the video codecs supported by the trunk
  • T38: special parameters for t38 support and the maxdatagram
  • From number: adds an expression to set a dynamic cid number for outgoing calls based on the office number or on the value “Set” -> “cid number” on trunk Dialplan application; works only if the operator allows setting of a cid number for outgoing calls
  • From name: adds a regular expression to set a dynaic cid name for outgoing calls
  • Cid Header: adds an additional header to set the cid number
  • Cid Body: indicates the contents of the cid header
  • Incoming CID: setting of a specific SIP header from which Cid header must be retrieved. Available options: p-asserted-identity,from (default) / from
  • Privacy Header: setting of the privacy header content supported by the operator to perform anonymous calls via feature code “Hide number” (92 by default)
  • Diversion / History-Info Header: adds SIP header Diversion and History-info for outgoing INVITE to trunk in order to preserve the information on the original called number in case of call forwarding
    
    

  
  

  • Show original caller number: can be used by operators which support displaying any number as cid, allows displaying original caller number from trunk in case of transfers/mobility calls
  • Support Refer and Hold: allows the trunk to perform transfers and disables hold requests from trunk
  • Session Timer: enables the check of the session validity to avoid pending calls; if enabled, the value 360secs is used, if disabled – 7200 secs
  • Force static SSRC: forbids SSRC change in RTP sessions; this option is recommended for some SIP operators which do not playback RTP packets after change of SSRC

  • Rport: the rport mechanism changes the SIP routing behavior, so that responses can be received through a NAT even if private addresses are used in the SIP headers; available options: INVITE (default), REGISTER/INVITE, off 

    Warning
    Limitation: If PBX is behind NAT and uses trunk with rport REGISTER, INVITE, the remote side may drop calls after 30 seconds.

    
    

  • Registration Expiry (sec): sets the expiry time for outgoing register messages (default=600 sec / min=0)
  • Custom DNS Server: used to resolve the sip proxy domain name
  • Outbound proxy: all outgoing SIP messages are sent through the indicated host
  • NAT IP: enables the network connection IP
  • 100rel: enables reliable transmission of provisional messages
  • Transport: select the transport protocol to be used: UDP/ TCP/ TLS/ auto (dns ptr-srv) 
  • DTMF mode: allows you to select the DTMF transfer mode: info/ rfc2833/ inband and specify the payload
    
    

  
  

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• Add new line into /etc/callweaver/sip-general-custom.conf: register => 144?144:123456:”144″@10.168.0.144:5060~~0.6 (where 0.6 is q-value)

• Run the command:

  Code Block
  callweaver -rx “sip reload”

  
  

How to enable transcoding for web phone calls to trunks which do not support G711

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Warning
Important It’s not recommended to enable this feature, as it reduces call quality and generates useless load on CPU It must be enabled only if the operator doesn’t support g711a/u for some calls It can generate CPU overload and problems if too many calls use it; in this case it is recommended to use another operator which supports all the needed codecs ( g711a / g711u / g729)

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• Port X. Below you can set up parameters for a single port:
  • Pricelist: the pricelist to calculate the costs for calls through the trunk
  • Local area code: specify the area code
  • End point type: select between TE (Terminal Equipment) (connection to public ISDN) or NT (Network Termination) mode (connection to ISDN phones or other devices in TE mode, e.g. PBXs)
  • Dialplan: Dialplan procedure for routing calls coming from this trunk (usually, “main”)
  • Clock: select between Generate clock and Use provided clock mode; in ISDN network one endpoint acts as a SERVERr, generating the clock signal, and the other endpoints act as CLIENTs, synchronizing on the clock signal received from the SERVER; typically a NT type endpoint acts as clock SERVER, and a TE type endpoint acts as clock CLIENT; set to CLIENT for connection to operators
  • Line Coding: select one of the line coding schemes: B8ZS (Bipolar 8-Zero Substitution), HDB3 (High Density Bipolar 3), AMI (Alternate Mark Inversion)
  • Network type: select the type of the ISDN network
  • Connection type: select the type of ISDN connection between point-to-point and point-to-multipoint
  • Link Establishment: select the connection establishment type between permanent and on demand
  • Signaling Protocol: select the signaling protocol between asymmetric DSS1 (used for connection to public ISDN) and symmetric QSIG (used for trunking between several PBXs)
  • Cid Name via Facility IE (previously “Additional services”): enables the gateway to accept Facility messages and enables calling name delivery for US and Canada

  • Overlap dialing (enabled by default): enables the gateway to begin processing a call as soon as it can determine a destination from dialed digits that form only part of a complete number; the feature must be enabled for DID (Direct Inward Dialing) management

    Warning
    Important: The option may need to be disabled for some VoIP Carriers for make outgoing calls due to the differences in D channel signaling performed for this option.

    
    

  • Inband DTMF Dialing: enables recognition of inband DTMF tones
  • Send Restart On Startup: enables sending of restart request to the operator each time the gateway reboots
  • Calling Name Max Length: specify the max length of the calling name
    
    

  
  

  • Channel Allocation Strategy: select the channel allocation strategy for this port
  • Sending Complete: includes Sending Complete information element into setup message; in this case the remote endpoint does not wait for digits coming in overlap mode
  • Progress Indicator: enables the gateway to provide the tone of progress during the indicated operations; normally these parameters are disabled
  • Maximum Facility Waiting Delay (ms): maximum waiting time of a Facility request
  • Use Implicit Inband Info: enables use of tones coming from endpoints

  • Correct incoming calling numbers: used for correct caller number visualization (adds 0 to national and 00 to international calls)

    Note
    Note: Starting from WMS Version 3.88.41047.24, the option is no longer applied for trunks with the USA country code: 0/ 00 are not added to national/ international incoming calls.

    
    

  • Signal Information Element: enables the use of Signal Information Element field to provide tones during different stages of call processing; this parameter must be enabled in NT mode only
  • Enable CNG / CED Tone Detection: allows voice and fax calls on the same line; enables the gateway to detect calling tone (CNG) generated by a fax machine, and called (answering) tone (CED) to enable T38 protocol

  • Detection Threshold: set up the threshold level for FAX detection

  
  

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